* Do a git blame on the embedded XML application or function element.
* From the commit hash, grab the summary line.
* Do a git log --grep <summary> to find the cherry-pick commits in all
branches that match.
* Do a git patch-id to ensure the commits are all related and didn't get
a false match on the summary.
* Do a git tag --contains <commit> to find the tags that contain each
commit.
* Weed out all tags not ..0.
* Sort and discard any .0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the application or function
was defined.
The applications and functions defined in the following files were done by
hand because the XML was extracted from the C source file relatively recently.
* channels/pjsip/dialplan_functions_doc.xml
* main/logger_doc.xml
* main/manager_doc.xml
* res/res_geolocation/geoloc_doc.xml
* res/res_stir_shaken/stir_shaken_doc.xml
* Do a git blame on the embedded XML managerEvent elements.
* From the commit hash, grab the summary line.
* Do a git log --grep <summary> to find the cherry-pick commits in all
branches that match.
* Do a git patch-id to ensure the commits are all related and didn't get
a false match on the summary.
* Do a git tag --contains <commit> to find the tags that contain each
commit.
* Weed out all tags not ..0.
* Sort and discard any .0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the application or function
was defined.
The events defined in res/res_pjsip/pjsip_manager.xml were done by hand
because the XML was extracted from the C source file relatively recently.
Two bugs were fixed along the way...
* The get_documentation awk script was exiting after it processed the first
DOCUMENTATION block it found in a file. We have at least 1 source file
with multiple DOCUMENTATION blocks so only the first one in them was being
processed. The awk script was changed to continue searching rather
than exiting after the first block.
* Fixing the awk script revealed an issue in logger.c where the third
DOCUMENTATION block contained a XML fragment that consisted only of
a managerEventInstance element that wasn't wrapped in a managerEvent
element. Since logger_doc.xml already existed, the remaining fragments
in logger.c were moved to it and properly organized.
Most of the configObjects and configOptions that are implemented with
ACO or Sorcery now have `<since>/<version>` elements added. There are
probably some that the script I used didn't catch. The version tags were
determined by the following...
* Do a git blame on the API call that created the object or option.
* From the commit hash, grab the summary line.
* Do a `git log --grep <summary>` to find the cherry-pick commits in all
branches that match.
* Do a `git patch-id` to ensure the commits are all related and didn't get
a false match on the summary.
* Do a `git tag --contains <commit>` to find the tags that contain each
commit.
* Weed out all tags not <major>.<minor>.0.
* Sort and discard any <major>.0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the API was last touched.
configObjects and configOptions elements implemented with the base
ast_config APIs were just not possible to find due to the non-deterministic
way they are accessed.
Also note that if the API call was on modified after it was added, the
version will be the one it was last modified in.
Final note: The configObject and configOption elements were introduced in
12.0.0 so options created before then may not have any XML documentation.
Correct an issue in ast_config_text_file_save2() when updating configuration
files with "#tryinclude" statements. The API currently replaces "#tryinclude"
with "#include". The API also creates empty template files if the referenced
files do not exist. This change resolves these problems.
Resolves: https://github.com/asterisk/asterisk/issues/920
* channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element.
* main/bucket.c: Removed XML completely since the "bucket" and "file" objects
are internal only with no config file.
* main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf"
and should have been "acl.conf"
* res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element.
* res/res_http_media_cache.c: Fixed the configFile element name. It was
"http_media_cache.conf" and should have been "res_http_media_cache.conf".
Essentially, we were treating 1234x1234 and 1234x5678 as 'equal'
because we were able to convert the prefix of each of these strings to
the same number.
Resolves: #1028
* Added the "since" element to the XML configObject and configOption elements
in appdocsxml.dtd.
* Added the "Since" section to the following CLI output:
```
config show help <module> <object>
config show help <module> <object> <option>
core show application <app>
core show function <func>
manager show command <command>
manager show event <event>
agi show commands topic <topic>
```
* Refactored the commands above to output their sections in the same order:
Synopsis, Since, Description, Syntax, Arguments, SeeAlso
* Refactored the commands above so they all use the same pattern for writing
the output to the CLI.
* Fixed several memory leaks caused by failure to free temporary output
buffers.
* Added a "since" array to the mustache template for the top-level resources
(Channel, Endpoint, etc.) and to the paths/methods underneath them. These
will be added to the generated markdown if present.
Example:
```
"resourcePath": "/api-docs/channels.{format}",
"requiresModules": [
"res_stasis_answer",
"res_stasis_playback",
"res_stasis_recording",
"res_stasis_snoop"
],
"since": [
"18.0.0",
"21.0.0"
],
"apis": [
{
"path": "/channels",
"description": "Active channels",
"operations": [
{
"httpMethod": "GET",
"since": [
"18.6.0",
"21.8.0"
],
"summary": "List all active channels in Asterisk.",
"nickname": "list",
"responseClass": "List[Channel]"
},
```
NOTE: No versioning information is actually added in this commit.
Those will be added separately and instructions for adding and maintaining
them will be published on the documentation site at a later date.
Configurations loaded with the ast_config_load2() API and later written
out with ast_config_text_file_save2() will have any leading whitespace
stripped away. The APIs should make reasonable efforts to maintain the
content and formatting of the configuration files.
This change retains any leading whitespace from comment lines that start
with a ";".
Resolves: https://github.com/asterisk/asterisk/issues/970
When using the ListCategories AMI action, it was possible to traverse
upwards through the directories to files outside of the configured
configuration directory. This action is now restricted to the configured
directory and an error will now be returned if the specified file is
outside of this limitation.
Resolves: #GHSA-33x6-fj46-6rfh
UserNote: The ListCategories AMI action now restricts files to the
configured configuration directory.
This was identified and fixed by @Allan-N in #918 but it is an
important fix in its own right.
The fix here is slightly different than Allan's in that we just move
the initialization of the problematic AO2 container to where it is
first used.
Fixes#1046
Previously, when AST_CONTROL_RINGING was received by
a DAHDI device, it would set its channel state to
AST_STATE_RINGING. However, an analysis of the codebase
and other channel drivers reveals RINGING corresponds to
physical power ringing, whereas AST_STATE_RING should be
used for audible ringback on the channel. This also ensures
the correct device state is returned by the channel state
to device state conversion.
Since there seems to be confusion in various places regarding
AST_STATE_RING vs. AST_STATE_RINGING, some documentation has
been added or corrected to clarify the actual purposes of these
two channel states, and the associated device state mapping.
An edge case that prompted this fix, but isn't explicitly
addressed here, is that of an incoming call to an FXO port.
The channel state will be "Ring", which maps to a device state
of "In Use", not "Ringing" as would be more intuitive. However,
this is semantic, since technically, Asterisk is treating this
the same as any other incoming call, and so "Ring" is the
semantic state (put another way, Asterisk isn't ringing anything,
like in the cases where channels are in the "Ringing" state).
Since FXO ports don't currently support Call Waiting, a suitable
workaround for the above would be to ignore the device state and
instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state)))
since it will be Ring if the FXO port is idle (but a call is ringing
on it) and Up if the FXO port is actually in use. (In both cases,
the device state would misleadingly be "In Use".)
Resolves: #1029
Unit tests can now be passed custom arguments from the command
line. For example, the following command would run the "mytest" test
in the "/main/mycat" category with the option "myoption=54"
`CLI> test execute category /main/mycat name mytest options myoption=54`
You can also pass options to an entire category...
`CLI> test execute category /main/mycat options myoption=54`
Basically, everything after the "options" keyword is passed verbatim to
the test which must decide what to do with it.
* A new API ast_test_get_cli_args() was created to give the tests access to
the cli_args->argc and cli_args->argv elements.
* Although not needed for the option processing, a new macro
ast_test_validate_cleanup_custom() was added to test.h that allows you
to specify a custom error message instead of just "Condition failed".
* The test_skel.c was updated to demonstrate parsing options and the use
of the ast_test_validate_cleanup_custom() macro.
Although C++ files (as extension .cc) have been handled in the module
directories for many years, the main directory was missing one line in its
Makefile that prevented C++ files from being recognised there.
ast_variable_retrieve currently returns the first match
for a variable, as opposed to the last one. This is problematic
because modules that load config settings by explicitly
calling ast_variable_retrieve on a variable name (as opposed
to iterating through all the directives as specified) will
end up taking the first specified value, such as the default
value from the template rather than the actual effective value
in an individual config section, leading to the wrong config.
This fixes this by making ast_variable_retrieve return the last
match, or the most recently overridden one, as the effective setting.
This is similar to what the -1 index in the AST_CONFIG function does.
There is another function, ast_variable_find_last_in_list, that does
something similar. However, it's a slightly different API, and it
sees virtually no usage in Asterisk. ast_variable_retrieve is what
most things use so this is currently the relevant point of breakage.
In practice, this is unlikely to cause any breakage, since there
would be no logical reason to use an inherited value rather than
an explicitly overridden value when loading a config.
ASTERISK-30370 #close
Resolves: #244
UpgradeNote: Config variables retrieved explicitly by name now return
the most recently overriding value as opposed to the base value (e.g.
from a template). This is equivalent to retrieving a config setting
using the -1 index to the AST_CONFIG function. The major implication of
this is that modules processing configs by explicitly retrieving variables
by name will now get the effective value of a variable as overridden in
a config rather than the first-set value (from a template), which is
consistent with how other modules load config settings.
When there is 0% packet loss on one side of the call and 15% packet loss on the other side, reading frame is often failed when reading direction_both audiohook. when read_factory available = 0, write_factory available = 320; i think write factory is usable read; because after reading one frame, there is still another frame that can be read together with the next read factory frame.
Resolves: #851
This update adds the processed call count to the CoreStatus AMI Action responsie. This output is
similar to the values returned by "core show channels" or "core show calls" in the CLI.
UserNote: The current processed call count is now returned as CoreProcessedCalls from the
CoreStatus AMI Action.
ast_variable_update currently sets the first match for a variable, as
opposed to the last one. This issue is complementary to that raised
in #244.
This is incorrect and results in the wrong (or no) action being taken
in cases where a section inherits from a template section. When the
traversal occurs to update the setting, the existing code erroneously
would use the first of possibly multiple matches in its update logic,
which is wrong. Now, explicitly use the last match in the traversal,
which will ensure that the actual setting is updated properly, and
not skipped or ignored because a template from which the setting's
section inherits was used for comparison.
Resolves: #960
UpgradeNote: Config variables, when set/updated, such as via AMI,
will now have the corresponding setting updated, even if their
sections inherit from template sections.
When the channel tech is multistream capable, the reference to
chan_topology was passed to the new channel. When the channel tech
isn't multistream capable, the reference to chan_topology was never
released. "Local" channels are multistream capable so it didn't
affect them but the confbridge "CBAnn" and the bridge_media
"Recorder" channels are not so they caused a leak every time one
of them was created.
Also added tracing to ast_stream_topology_alloc() and
stream_topology_destroy() to assist with debugging.
Resolves: #938
The dnsmgr_refresh() function checks to see if the IP address associated
with a name/service has changed. The gotcha is that the ast_get_ip_or_srv()
function only returns the first IP address returned by the DNS query. If
there are multiple IPs associated with the name and the returned order is
not consistent (e.g. with DNS round-robin) then the other IP addresses are
not included in the comparison and the entry is flagged as changed even
though the IP is still valid.
Updated the code to check all IP addresses and flag a change only if the
original IP is no longer valid.
Resolves: #924
Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be
balanced to ensure that we can capture the exit status of child
processes when we need to. This extends to functions that call
`ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as
`ast_safe_fork()` and `ast_safe_fork_cleanup()`.
The primary change here is ensuring that we do not call
`ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously
called `ast_safe_fork()`.
Additionally we reinforce some of the documentation and add an
assertion to, ideally, catch this sooner were this to happen again.
Fixes#922
asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings.
config.c: Include header file for WIFEXITED/WEXITSTATUS macros.
res_timing_kqueue: Use more portable format specifier.
test_crypto: Use non-linux limits.h header file.
Resolves: #916
When using the ModuleLoad AMI action, it was possible to traverse
upwards through the directories to files outside of the configured
modules directory. We decided it would be best to restrict access to
modules exclusively in the configured directory. You will now get an
error when the specified module is outside of this limitation.
Fixes: #897
UserNote: The ModuleLoad AMI action now restricts modules to the
configured modules directory.
Don't pass through a NULL argument to fclose, which is undefined
behavior, and instead return -1 and set errno appropriately. This
also avoids a compiler warning with glibc 2.38 and newer, as glibc
commit 71d9e0fe766a3c22a730995b9d024960970670af
added the nonnull attribute to this argument.
Resolves: #900
In certain circumstances a channel may undergo an operation
referred to as a masquerade. If this occurs the CHANNEL(userfield)
value was not preserved causing it to get lost. This change makes
it so that this field is now preserved.
Fixes: #882
The app is actually named "BackGround" but several references
in XML documentation were spelled "Background" with the lower
case "g". This was causing documentation links to return
"not found" messages.
UserNote: You can now perform more granular filtering on events
in manager.conf using expressions like
`eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
This is much more efficient than
`eventfilter = Event: Newchannel.*Channel: PJSIP/`
Full syntax guide is in configs/samples/manager.conf.sample.
Consumers like media_cache have been running into issues with
the previous astdb "/family/key" limit of 253 bytes when needing
to store things like long URIs. An Amazon S3 URI is a good example
of this. Now, instead of using a static 256 byte buffer for
"/family/key", we use ast_asprintf() to dynamically create it.
Both test_db.c and test_media_cache.c were also updated to use
keys/URIs over the old 253 character limit.
Resolves: #881
UserNote: The `ast_db_*()` APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the `DB*` dialplan applications, dialplan functions,
manager actions and `databse` CLI commands. Since the
media_cache also uses the `ast_db_*()` APIs, you can now store
resources with URIs longer than 253 bytes.
It's possible that ast_autoservice_stop is called within the autoservice thread.
In this case the autoservice thread is stuck in an endless sleep.
To avoid endless sleep ast_autoservice_stop must check that it's not called
within the autoservice thread.
Fixes: #763
The ub_result pointer passed to unbound_resolver_callback by
libunbound can be NULL if the query was for something malformed
like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result
and return instead of crashing with a SEGV. This causes pjproject
to simply cancel the transaction with a "No answer record in the DNS
response" error. The existing "off nominal" unit test was also
updated to check this condition.
Although not necessary for this fix, we also made
ast_dns_resolver_completed() tolerant of a NULL result.
Resolves: GHSA-v428-g3cw-7hv9
When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
matching 4733 offers, Bob may want to use the 48K audio codec but can not
accept 48K digits and so negotiates for a mixed set.
Asterisk will now check Bob's offer to make sure Bob has indicated this is
acceptible and if not, will use Bob's preference.
Fixes: #847
PR #700 added a preferred_format for the struct ast_rtp_codecs,
but when set the preferred_format it leaks an astobj2 ast_format.
In the next code
ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0));
both functions ast_rtp_codecs_set_preferred_format
and ast_format_cap_get_format increases the ao2 reference count.
Fixes: #856
If you run an AMI CoreShowChannelMap on a channel that isn't in a
bridge and you're in DEVMODE, you can get a FRACK because the
bridge id is empty. We now simply return an empty list for that
request.
This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)
It can also be accessed via CHANNEL:
exten => example,2,NoOp(CHANNEL(tenantid))
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
A stasis event is now produced when using the TONE_DETECT dialplan
function. This event is published over ARI using the ChannelToneDetected
event. This change does not make it available over AMI.
Fixes: #811
UserNote: Setting the TONE_DETECT dialplan function on a channel
in ARI will now cause a ChannelToneDetected ARI event to be raised
when the specified tone is detected.
Added Reload and DBdeltree to the list of dialplan application that
can't be executed via the Originate manager action without also
having write SYSTEM permissions.
Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan
functions that can't be executed via the Originate manager action
without also having write SYSTEM permissions.
If the Queue application is attempted to be run by the Originate
manager action and an AGI parameter is specified in the app data,
it'll be rejected unless the manager user has either the AGI or
SYSTEM permissions.
Resolves: #GHSA-c4cg-9275-6w44
There can be empty slots in payload_mapping_tx corresponding to
dynamic payload types that haven't been seen before so we now
check for NULL before attempting to use 'type' in the call to
ast_format_cmp.
Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
Resolves: #822
Whenver a new channel snapshot is created or when a channel is
destroyed, we need to delete any existing channel snapshot from
the snapshot cache. Historically, we used the channel->snapshot
pointer to delete any existing snapshots but this has two issues.
First, if something (possibly ast_channel_internal_swap_snapshots)
sets channel->snapshot to NULL while there's still a snapshot in
the cache, we wouldn't be able to delete it and it would be orphaned
when the channel is destroyed. Since we use the cache to list
channels from the CLI, AMI and ARI, it would appear as though the
channel was still there when it wasn't.
Second, since there are actually two caches, one indexed by the
channel's uniqueid, and another indexed by the channel's name,
deleting from the caches by pointer requires a sequential search of
all of the hash table buckets in BOTH caches to find the matching
snapshots. Not very efficient.
So, we now delete from the caches using the channel's uniqueid
and name. This solves both issues.
This doesn't address how channel->snapshot might have been set
to NULL in the first place because although we have concrete
evidence that it's happening, we haven't been able to reproduce it.
Resolves: #783
This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
which when set causes the ODBC variant of app_voicemail to leave
the message and greeting audio files on disk and only store the
message metadata in the database. This option came from a concern
that the database could grow to large and cause remote access
and/or replication to become slow. In a clustering situation
with this option, all asterisk instances would share the same
database for the metadata and either use a shared filesystem
or other filesystem replication service much more suitable
for synchronizing files.
The changes to app_voicemail to implement this feature were actually
quite small but due to the complexity of the module, the actual
source code changes were greater. They fall into the following
categories:
* Tracing. The module is so complex that it was impossible to
figure out the path taken for various scenarios without the addition
of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
code that's not related to the functional change. Making this worse
was the fact that many "if" statements in this module didn't use
braces. Since the tracing macros add multiple statements, many "if"
statements had to be converted to use braces.
* Excessive use of PATH_MAX. Previous maintainers of this module
used PATH_MAX to allocate character arrays for filesystem paths
and SQL statements as though they cost nothing. In fact, PATH_MAX
is defined as 4096 bytes! Some functions had (and still have)
multiples of these. One function has 7. Given that the vast
majority of installations use the default spool directory path
`/var/spool/asterisk/voicemail`, the actual path length is usually
less than 80 bytes. That's over 4000 bytes wasted. It was the
same for SQL statement buffers. A 4K buffer for statement that
only needed 60 bytes. All of these PATH_MAX allocations in the
ODBC related code were changed to dynamically allocated buffers.
The rest will have to be addressed separately.
* Bug fixes. During the development of this feature, several
pre-existing ODBC related bugs were discovered and fixed. They
had to do with leaving orphaned files on disk, not preserving
original message ids when moving messages between folders,
not honoring the "formats" config parameter in certain circumstances,
etc.
UserNote: This commit adds a new voicemail.conf option
'odbc_audio_on_disk' which when set causes the ODBC variant of
app_voicemail_odbc to leave the message and greeting audio files
on disk and only store the message metadata in the database.
Much more information can be found in the voicemail.conf.sample
file.
From the gdb information, we can see that while iterating over bridge->channels, the ast_bridge_channel reference count is 0, indicating it has already been destroyed.Additionally, when ast_bridge_channel is removed from bridge->channels, the bridge is first locked. Therefore, locking the bridge before iterating over bridge->channels can resolve the race condition.
Resolves: https://github.com/asterisk/asterisk/issues/768
The current width for "extension" is 20 and "device state id" is 20, which is too small.
The "extension" field contains "ext"@"context", so 20 characters is not enough.
The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.
Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.
Resolves: #770
UserNote: The fields width of "core show hints" were increased.
The width of "extension" field to 30 characters and
the width of the "device state id" field to 60 characters.
Since Asterisk 19 it is possible to cache recorded files into another
directory [1] [2].
Show configured location of cache dir in CLI's core show settings.
[1] ASTERISK-29143
[2] b08427134f
Two functions are deprecated as of libxml2 2.12:
* xmlSubstituteEntitiesDefault
* xmlParseMemory
So we update those with supported API.
Additionally, `res_calendar_caldav` has been updated to use libxml2's
xmlreader API instead of the SAX2 API which has always felt a little
hacky (see deleted comment block in `res_calendar_caldav.c`).
The xmlreader API has been around since libxml2 2.5.0 which was
released in 2003.
Fixes#725
Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.
Asterisk currently treats RFC2833 Digits as a single rtp payload type
with a fixed bitrate of 8K. This change would expand that to 8, 16,
24 and 32K.
This requires checking the offered rtp types for any of these bitrates
and then adding an offer for each (if configured for RFC2833.) DTMF
generation must also be changed in order to look at the current outbound
codec in order to generate appropriately timed rtp.
For cases where no outgoing audio has yet been sent prior to digit
generation, Asterisk now has a concept of a 'preferred' codec based on
offer order.
On inbound calls Asterisk will mimic the payload types of the RFC2833
digits.
On outbound calls Asterisk will choose the next free payload types starting
with 101.
UserNote: No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Resolves: #699
Send "RELOADING=1" instead of "RELOAD=1" to follow the format
expected by systemd (see sd_notify(3) man page).
Do not send STOPPING=1 in remote console mode:
attempting to execute "asterisk -rx" by the main process leads to
a warning if NotifyAccess=main (the default) or to a forced termination
if NotifyAccess=all.
Some of the money announcements can be off by one cent,
due to the use of floating point in the money calculations,
which is bad for obvious reasons.
This replaces floating point with simple string parsing
to ensure the cents value is converted accurately.
Resolves: #525
Because of the (often recursive) nature of module dependencies in
Asterisk, hot swapping a module on the fly is cumbersome if a module
is depended on by other modules. Currently, dependencies must be
popped manually by unloading dependents, unloading the module of
interest, and then loading modules again in reverse order.
To make this easier, the ability to do this recursively in certain
circumstances has been added, as an optional extension to the
"module refresh" command. If requested, Asterisk will check if a module
that has a positive usecount could be unloaded safely if anything
recursively dependent on it were unloaded. If so, it will go ahead
and unload all these modules and load them back again. This makes
hot swapping modules that provide dependencies much easier.
Resolves: #474
UserNote: In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
If the hostname field of the ast_tcptls_session_args structure is
set (which it is for websocket client connections), that hostname
will now automatically be used in an SNI TLS extension in the client
hello.
Resolves: #713
UserNote: Secure websocket client connections now send SNI in
the TLS client hello.
Commit f2f397c1a8 previously
made it possible to send Caller ID parameters to FXS stations
which, prior to that, could not be sent.
This change is complementary in that we now handle receiving
all these parameters on FXO lines and provide these up to
the dialplan, via chan_dahdi. In particular:
* If a redirecting reason is provided, the channel's redirecting
reason is set. No redirecting number is set, since there is
no parameter for this in the Caller ID protocol, but the reason
can be checked to determine if and why a call was forwarded.
* If the Call Qualifier parameter is received, the Call Qualifier
variable is set.
* Some comments have been added to explain why some of the code
is the way it is, to assist other people looking at it.
With this change, Asterisk's Caller ID implementation is now
reasonably complete for both FXS and FXO operation.
Resolves: #681
rtp_engine.c and stun.c were calling ast_register_cleanup which
is skipped if any loadable module can't be cleanly unloaded
when asterisk shuts down. Since this will always be the case,
their cleanup functions never get run. In a practical sense
this makes no difference since asterisk is shutting down but if
you're in development mode and trying to use the leak sanitizer,
the leaks from both of those modules clutter up the output.
There were a few references in the embedded documentation XML
where the case didn't match or where the referenced app or function
simply didn't exist any more. These were causing 404 responses
in docs.asterisk.org.
manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too.
For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient.
UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
It is possible for dialplan to result in an infinite
recursion of variable substitution, which eventually
leads to stack overflow. If we detect this, abort
substitution and log an error for the user to fix
the broken dialplan.
Resolves: #480
UpgradeNote: The maximum amount of dialplan recursion
using variable substitution (such as by using EVAL_EXTEN)
is capped at 15.
This adds a CLI command that can be used to manually
kick specific AMI sessions.
Resolves: #485
UserNote: The "manager kick session" CLI command now
allows kicking a specified AMI session.
Why do we need a refactor?
The original stir/shaken implementation was started over 3 years ago
when little was understood about practical implementation. The
result was an implementation that wouldn't actually interoperate
with any other stir-shaken implementations.
There were also a number of stir-shaken features and RFC
requirements that were never implemented such as TNAuthList
certificate validation, sending Reason headers in SIP responses
when verification failed but we wished to continue the call, and
the ability to send Media Key(mky) grants in the Identity header
when the call involved DTLS.
Finally, there were some performance concerns around outgoing
calls and selection of the correct certificate and private key.
The configuration was keyed by an arbitrary name which meant that
for every outgoing call, we had to scan the entire list of
configured TNs to find the correct cert to use. With only a few
TNs configured, this wasn't an issue but if you have a thousand,
it could be.
What's changed?
* Configuration objects have been refactored to be clearer about
their uses and to fix issues.
* The "general" object was renamed to "verification" since it
contains parameters specific to the incoming verification
process. It also never handled ca_path and crl_path
correctly.
* A new "attestation" object was added that controls the
outgoing attestation process. It sets default certificates,
keys, etc.
* The "certificate" object was renamed to "tn" and had it's key
change to telephone number since outgoing call attestation
needs to look up certificates by telephone number.
* The "profile" object had more parameters added to it that can
override default parameters specified in the "attestation"
and "verification" objects.
* The "store" object was removed altogther as it was never
implemented.
* We now use libjwt to create outgoing Identity headers and to
parse and validate signatures on incoming Identiy headers. Our
previous custom implementation was much of the source of the
interoperability issues.
* General code cleanup and refactor.
* Moved things to better places.
* Separated some of the complex functions to smaller ones.
* Using context objects rather than passing tons of parameters
in function calls.
* Removed some complexity and unneeded encapsuation from the
config objects.
Resolves: #351Resolves: #46
UserNote: Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
UpgradeNote: The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio.
This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).
- add new table mode
- hide the 999999 comp values, as these only indicate an issue with transcoding
- hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding)
Resolves: #601
If ast_dsp_process is called with a codec besides slin, ulaw,
or alaw, a warning is logged that in-band DTMF is not supported,
but this message is not always appropriate or correct, because
ast_dsp_process is much more generic than just DTMF detection.
This logs a more generic message in those cases, and also improves
codec-mismatch logging throughout dsp.c by ensuring incompatible
codecs are printed out.
Resolves: #595
Under rare circumstances, it's possible for the original audio
session in the active_media_state default_session to be corrupted
instead of removed when switching to the t38/image media session
during fax negotiation. This can cause a segfault when a "pjsip
show channelstats" attempts to print that audio media session's
rtp statistics. In these cases, the active_media_state
topology is correctly showing only a single t38/image stream
so we now check that there's an audio stream in the topology
before attempting to use the audio media session to get the rtp
statistics.
Resolves: #592
When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering. Besides taking up
resources, it also makes it hard to debug failing tests.
This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.
There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.
Resolves: #582
Given the scenario of passing an empty string to the
ast_strsep functions the functions would return NULL
instead of an empty string. This is counter to how
strsep itself works.
This change alters the behavior of the functions to
match that of strsep.
Fixes: #565
Currently, a reload will always occur if the
Reload header is provided for the UpdateConfig
action. However, we should not be doing a reload
if the header value has a falsy value, per the
documentation, so this makes the reload behavior
consistent with the existing documentation.
Resolves: #551
Commit 008731b0a4
caused a regression by resulting in logger.xml
being compiled and linked into the asterisk
binary in lieu of logger.c on certain platforms
if Asterisk was compiled in dev mode.
To fix this, we ensure the file has a unique
name without the extension. Most existing .xml
files have been named differently from any
.c files in the same directory or did not
pose this issue.
channels/pjsip/dialplan_functions.xml does not
pose this issue but is also being renamed
to adhere to this policy.
Resolves: #539
This adds a simple CLI command that can be used for
analyzing all frames currently queued to a channel.
A couple log messages are also adjusted to be more
useful in tracing bridging problems.
Resolves: #533
This reverts commit 315eb551db.
Over the past year, we've had several reports of "topology storms"
occurring where 2 external facing channels connected by one or more
local channels and bridges will get themselves in a state where
they continually send each other topology change requests. This
usually manifests itself in no-audio calls and a flood of
"Exceptionally long queue length" messages. It appears that this
commit is the cause so we're reverting it for now until we can
determine a more appropriate solution.
Resolves: #530
Commit 424be34563 introduced
a regression by calling ast_free on memory allocated by
realpath. This causes Asterisk to abort when executing this
function. Since the memory is allocated by glibc, it should
be freed using ast_std_free.
Resolves: #513
When using AMI GetConfig, it was possible to access files outside of the
Asterisk configuration directory by using filenames with ".." and "./"
even while live_dangerously was not enabled. This change resolves the
full path and ensures we are still in the configuration directory before
attempting to access the file.
Improve the "manager show connected" CLI command
to clarify that the last two columns are permissions
related, not counts, and use sufficient widths
to consistently display these values.
ASTERISK-30143 #close
Resolves: #482
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36.
Resolves: #430
The `get_documentation` awk script will only extract the first
DOCUMENTATION block that it finds in a given file. This is by design
(9bc2127) to prevent AMI event documentation from being pulled in to
the core.xml documentation file.
Because of this, the `LOG_GROUP` documentation added in 89709e2 was
not being properly extracted and was missing fom the resulting XML
documentation file. This commit moves the `LOG_GROUP` documentation to
a separate `logger.xml` file.
If the script referenced by `#exec` does not exist, writes anything to
stderr, or exits abnormally or with a non-zero exit status, we log
that to Asterisk's error logging channel.
Additionally, write out a warning if the script produces no output.
Fixes#259
Any function or application that accepts a `&`-separated list of
filenames can now include a literal `&` in a filename by wrapping the
entire filename in single quotes, e.g.:
```
exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world)
```
Fixes#172
UpgradeNote: Ampersands in URLs passed to the `Playback()`,
`Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
`Queue()` applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the `CONFBRIDGE` dialplan function, or configuring various
features in `confbridge.conf` and `queues.conf`.
Fixes a crash due to a lack of proper reference on the nativeformats
object before passing it into ast_request(). Also found potentially
similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c
Fixes: #388
This adds the ability to filter console
logging by channel or groups of channels.
This can be useful on busy systems where
an administrator would like to analyze certain
calls in detail. A dialplan function is also
included for the purpose of assigning a channel
to a group (e.g. by tenant, or some other metric).
ASTERISK-30483 #close
Resolves: #242
UserNote: The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
Currently, trying to call a Local channel with a slash
in the extension will fail due to the parsing of characters
after such a slash as being dial modifiers. Additionally,
core_local is inconsistent and incomplete with
its parsing of Local dial strings in that sometimes it
uses the first slash and at other times it uses the last.
For instance, something like DAHDI/5 or PJSIP/device
is a perfectly usable extension in the dialplan, but Local
channels in particular prevent these from being called.
This creates inconsistent behavior for users, since using
a slash in an extension is perfectly acceptable, and using
a Goto to accomplish this works fine, but if specified
through a Local channel, the parsing prevents this.
This fixes this by explicitly parsing options from the
last slash in the extension, rather than the first one,
which doesn't cause an issue for extensions with slashes.
ASTERISK-30013 #close
Resolves: #248
This commit fixes crashes in JSON_DECODE() for types null, true, false
and real numbers.
In addition it ensures that a path is not deeper than 32 levels.
Also allow root object to be an array.
Add unit tests for above cases.
Some providers require a multiple of 20 for the maxptime or fail to complete calls,
e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.
Resolves: #260
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds. From a code perspective, the only reason they were
tied together was for logging. So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.
Resolves: #321
UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
The CLI .asterisk_history file is read from/written to the directory
specified by the HOME environment variable. If the root user starts
asterisk with the -U/-G options, or with runuser/rungroup set in
asterisk.conf, the asterisk process is started as root but then it
calls setuid/setgid to set the new user/group. This does NOT reset
the HOME environment variable to the new user's home directory
though so it's still left as "/root". In this case, the new user
will almost certainly NOT have access to read from or write to the
history file.
* Added function process_histfile() which calls
getpwuid(geteuid()) and uses pw->dir as the home directory
instead of the HOME environment variable.
* ast_el_read_default_histfile() and ast_el_write_default_histfile()
have been modified to use the new process_histfile()
function.
Resolves: #337
To better co-exist with sounds files that may be managed by
packages, custom sound files may now be placed in
AST_DATA_DIR/sounds/custom instead of the standard
AST_DATA_DIR/sounds/<lang> directory. If the new
"sounds_search_custom_dir" option in asterisk.conf is set
to "true", asterisk will search the custom directory for sounds
files before searching the standard directory. For performance
reasons, the "sounds_search_custom_dir" defaults to "false".
Resolves: #315
UserNote: A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/<lang> directory.
The previous behavior of make_buildopts_h was to not add the
non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
REF_DEBUG, etc. to the buildopts.h file because "it caused
ccache to invalidate files and extended compile times". They're
only defined by passing them on the gcc command line with '-D'
options. In practice, including them in the include file rarely
causes any impact because the only time ccache cares is if you
actually change an option so the hit occurrs only once after
you change it.
OK so why would we want to include them? Many IDEs follow the
include files to resolve defines and if the options aren't in an
include file, it can cause the IDE to mark blocks of "ifdeffed"
code as unused when they're really not.
So...
* Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
which tells make_buildopts_h to include the non-ABI-breaking
flags in buildopts.h as well as the ABI-breaking ones. The default
is disabled to preserve current behavior. As before though,
only the ABI-breaking flags appear in AST_BUILDOPTS and only
those are used to calculate AST_BUILDOPT_SUM.
A new AST_BUILDOPT_ALL define was created to capture all of the
flags.
* make_version_c was streamlined to use buildopts.h and also to
create asterisk_build_opts_all[] and ast_get_build_opts_all(void)
* "core show settings" now shows both AST_BUILDOPTS and
AST_BUILDOPTS_ALL.
UserNote: The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.
Resolves: #71
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
Adds last locked and unlocked timestamps as well as a
counter for the number of times the lock has been
attempted (vs locked/unlocked) to debug output printed
using the DEBUG_THREADS option.
Resolves: #110
If we don't set this to -1 if the structure can be potentially re-used
later then it's possible that we'll issue a close() on an unrelated file
descriptor, breaking asterisk in other interesting ways.
I believe this to be an unlikely scenario, but it costs nothing to be
safe.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
* gcc 13 is now catching when a function is declared as returning
an enum but defined as returning an int or vice versa. Fixed
a few in app.h, loader.c, stasis_message.c.
* gcc 13 is also now (incorrectly) complaining of dangling pointers
when assigning a pointer to a local char array to a char *. Had
to change that to an ast_alloca.
Resolves: #155
Adds a new AMI action (CoreShowChannelMap) that takes in a channel name
and provides a list of all channels that are connected to that channel,
following local channel connections as well.
Resolves: #104
UserNote: New AMI action CoreShowChannelMap has been added.
The Caller ID generation routine currently is hardcoded
to always use the system time zone. This makes it possible
to optionally specify any TZ-format time zone.
Resolves: #98
ASTERISK-30330
Commit 09e989f972
categorized the T option as not being compatible
with remote consoles, but they do affect verbose
messages with remote console. This fixes this.
Resolves: #102
If processing an XInclude results in new <xi:include> elements, we
need to run XInclude processing again. This continues until no
replacement occurs or an error is encountered.
There is a separate issue with dynamic strings (ast_str) that will be
addressed separately.
Resolves: #65
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred. This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
Resolves: #52
UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
ast_waitstream was not called after ast_streamfile,
resulting in "o'clock" being skipped in French.
Additionally, the minute announcements should be
feminine.
Reported-by: Danny Lloyd
Resolves: #41
ASTERISK-30488
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.
RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.
To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.
Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.
ASTERISK-30407 #close
Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2
The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.
* The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
* The "failures" attribute was added to the "testsuite"
element.
Also added 2 new unit tests that just pass and fail to be
used for CI testing.
Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69
Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.
Some notes:
* Because of the version of flex we are using to generate the lexers
(2.5.35) some post-processing in the Makefile is still required.
* The generated lexers do not contain the problematic C99 check that
was being replaced by the call to sed in the respective Makefiles so
it was removed.
* Since these files are generated, they will include trailing
whitespace in some places. This does not need to be corrected.
Change-Id: Ibbd343606fcf5c0d285b1599e6e8e59f514f2e4e
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed. If the interval is set to an
invalid value, it will default to 15 seconds.
A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started. This is so we can test that the touch
variable starts the hook as expected.
ASTERISK-30446
Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.
This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute. This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.
ASTERISK-30464
Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98
For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
ASTERISK-30455
Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1
Change the HTTP status page (located at /httpstatus by default) by:
* Combining the address and port into a single line.
* Changing "SSL" to "TLS"
ASTERISK-30433 #close
Change-Id: Id2ccb1218f00a68424aca2b651647d8b1f549bcb
Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
d7ec516916:
iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
559 | if (!stream->ssl->server) {
| ^~
ASTERISK-30107 #close
Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
Change-Id: Iea7f34970297f2fb50285d73462d0174ba7e9587
* Added a new function ast_utf8_replace_invalid_chars() to
utf8.c that copies a string replacing any invalid UTF-8
sequences with the Unicode specified U+FFFD replacement
character. For example: "abc\xffdef" becomes "abc\uFFFDdef".
Any UTF-8 compliant implementation will show that character
as a � character.
* Updated res_pjsip:set_id_from_hdr() to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
* Updated stasis_channels:ast_channel_publish_varset to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
ASTERISK-27830
Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8
This avoids buffer overflow errors when running tests that capture
output from child processes.
This also corrects a copypasta in an off-nominal error message.
Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d
For most modules that interacted with app_macro, this change is limited
to no longer looking for the current context from the macrocontext when
set. Additionally, the following modules are impacted:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs a re-write
app_queue - can no longer a macro on the called party's channel.
Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected line or
redirection macro options
options - stdexten is deprecated to gosub as the default and only
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
ASTERISK-30304
Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6
-----------------
This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures. The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.
ASTERISK-30391
-----------------
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038
If native HTTP is disabled but HTTPS is enabled and status page enabled
too, Core/HTTP crashes while loading. 'global_http_server' references
to NULL, but the status page tries to dereference it.
The patch adds a check for HTTP is enabled.
ASTERISK-30379 #close
Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249
Currently, if a module declines to load, dlopen is called
to register the module but dlclose never gets called.
Furthermore, loader.c currently doesn't allow dlclose
to ever get called on the module, since it declined to
load and the unload function bails early in this case.
This can be problematic if a module is updated, since the
new module cannot be loaded into memory since we haven't
closed all references to it. To fix this, we now allow
modules to be unloaded, even if they never "loaded" in
Asterisk itself, so that dlclose is called and the module
can be properly cleaned up, allowing the updated module
to be loaded from scratch next time.
ASTERISK-30345 #close
Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5
The if statement here is always false after the for
loop finishes, so variables are never appended.
This removes that to properly append to the end
of the variable list.
ASTERISK-30351 #close
Reported by: Sebastian Gutierrez
Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
* Updated chan_pjsip to set quality channel variables when a
call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
to retrieve the MES along with the existing rtcp stats when
using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
requested. Also debug output that dumps the stats when an
rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
MES. In the process, also had to update the calculation of
jitter. Many debugging statements were also changed to be
more informative.
* Added a unit test for internal testing. The test should not be
run during normal operation and is disabled by default.
ASTERISK-30280
Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a
pbx_exec makes a channel snapshot before executing applications.
This doesn't cause an issue during normal dialplan execution
where pbx_exec is called over and over again in succession.
However, if pbx_exec is called "one off", e.g. using
ast_pbx_exec_application, then a channel snapshot never ends
up getting made after the executed application returns, and
inaccurate snapshot information will linger for a while, causing
"core show channels", etc. to show erroneous info.
This is fixed by manually making a channel snapshot at the end
of ast_pbx_exec_application, since we anticipate that pbx_exec
might not get called again immediately.
ASTERISK-30367 #close
Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086
This removes the ImportVar and SetAMAFlags applications
which have been deprecated since Asterisk 12, but were
never removed previously.
Additionally, it removes remnants of defunct options
that themselves were removed years ago.
ASTERISK-30335 #close
Change-Id: I749520c7b08d4c9d5eebbf640d4fbc81950eda8d
When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.
This change ensures that the stream is closed in case of error.
ASTERISK-30198 #close
Reported-by: Julien Alie
Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803
The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).
This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.
ASTERISK-30289 #close
Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07
Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.
ASTERISK-30176
Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404
(cherry picked from commit 81f10e847e)
The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.
This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.
ASTERISK-30308 #close
Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4
Sean Bright
George Joseph
Allan Nathanson
Steffen Arntz
Ben Ford
Naveen Albert
Tinet-mucw
Mike Pultz
James Terhune
Peter Jannesen
Alexei Gradinari
Mike Bradeen
gibbz00
Bastian Triller
Ivan Poddubny
Spiridonov Dmitry
jonatascalebe
Sebastian Jennen
Joshua C. Colp
Brad Smith
Matthew Fredrickson
Holger Hans Peter Freyther
Eduardo
Maximilian Fridrich
Jaco Kroon
InterLinked1
Sean Bright
Boris P. Korzun
Fabrice Fontaine
George Joseph
Peter Fern
Michael Kuron